16b6b90a82
In this CL I have rewritten the way we handle stereo packets in VoE. Before this change we split the packets in the RTP module and added two packets to ACM, one for the left channel and one for the right. This lead to timing problems caused when a different thread called RecOut in between the two calls to add stereo packet to ACM. (RecOut is called to pull audio data, decode packets, on the receiving side). While doing the change I also took the opportunity to changed some functions so that the data stream is uint8 everywhere. The list of files in this CL is long, but should be fairly easy to review. It is difficult to see what has been changed in some of the tests, but I can explain offline. Reviewers: Björn - /src/modules/audio_coding Patrik - /src/modules/rtp_rtcp Patrik -/src/modules/utility Henrik A - /src/voice_engine BUG=410 TEST=voe_cmd_test Review URL: https://webrtc-codereview.appspot.com/473003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2012 4adac7df-926f-26a2-2b94-8c16560cd09d
384 lines
11 KiB
C++
384 lines
11 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
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#include "audio_coding_module.h"
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#include "audio_coding_module_typedefs.h"
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#include "engine_configurations.h"
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#include "module_common_types.h"
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#include "typedefs.h"
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#include "webrtc_neteq.h"
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#include "webrtc_vad.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class RWLockWrapper;
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struct CodecInst;
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enum AudioPlayoutMode;
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enum ACMSpeechType;
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#define MAX_NUM_SLAVE_NETEQ 1
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class ACMNetEQ
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{
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public:
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// Constructor of the class
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ACMNetEQ();
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// Destructor of the class.
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~ACMNetEQ();
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//
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// GetVersion()
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// Fills the version array with the NetEQ version and updates the
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// remainingBufferInBytes and position variables accordingly.
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//
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// Output:
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// - version : An array to be filled with the version
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// data.
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//
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// Input/Output:
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// - remainingBuffInBytes : The number of free bytes at the end of
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// the version array.
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// - position : Position where the free space starts.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ returned an error.
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//
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static WebRtc_Word32 GetVersion(
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char* version,
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WebRtc_UWord32& remainingBuffInBytes,
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WebRtc_UWord32& position);
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//
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// Init()
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// Allocates memory for NetEQ and VAD and initializes them.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ or VAD returned an error or
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// if out of memory.
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//
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WebRtc_Word32 Init();
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//
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// RecIn()
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// Gives the payload to NetEQ.
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//
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// Input:
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// - incomingPayload : Incoming audio payload.
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// - payloadLength : Length of incoming audio payload.
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// - rtpInfo : RTP header for the incoming payload containing
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// information about payload type, sequence number,
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// timestamp, ssrc and marker bit.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 RecIn(
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const WebRtc_UWord8* incomingPayload,
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const WebRtc_Word32 payloadLength,
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const WebRtcRTPHeader& rtpInfo);
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//
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// RecOut()
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// Asks NetEQ for 10 ms of decoded audio.
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//
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// Input:
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// -audioFrame : an audio frame were output data and
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// associated parameters are written to.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ returned an error.
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//
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WebRtc_Word32 RecOut(
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AudioFrame& audioFrame);
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//
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// AddCodec()
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// Adds a new codec to the NetEQ codec database.
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//
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// Input:
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// - codecDef : The codec to be added.
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// - toMaster : true if the codec has to be added to Master
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// NetEq, otherwise will be added to the Slave
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// NetEQ.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 AddCodec(
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WebRtcNetEQ_CodecDef *codecDef,
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bool toMaster = true);
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//
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// AllocatePacketBuffer()
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// Allocates the NetEQ packet buffer.
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//
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// Input:
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// - usedCodecs : An array of the codecs to be used by NetEQ.
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// - noOfCodecs : Number of codecs in usedCodecs.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 AllocatePacketBuffer(
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const WebRtcNetEQDecoder* usedCodecs,
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WebRtc_Word16 noOfCodecs);
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//
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// SetExtraDelay()
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// Sets an delayInMS milliseconds extra delay in NetEQ.
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//
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// Input:
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// - delayInMS : Extra delay in milliseconds.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 SetExtraDelay(
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const WebRtc_Word32 delayInMS);
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//
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// SetAVTPlayout()
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// Enable/disable playout of AVT payloads.
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//
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// Input:
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// - enable : Enable if true, disable if false.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 SetAVTPlayout(
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const bool enable);
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//
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// AVTPlayout()
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// Get the current AVT playout state.
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//
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// Return value : True if AVT playout is enabled.
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// False if AVT playout is disabled.
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//
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bool AVTPlayout() const;
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//
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// CurrentSampFreqHz()
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// Get the current sampling frequency in Hz.
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//
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// Return value : Sampling frequency in Hz.
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//
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WebRtc_Word32 CurrentSampFreqHz() const;
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//
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// SetPlayoutMode()
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// Sets the playout mode to voice or fax.
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//
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// Input:
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// - mode : The playout mode to be used, voice,
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// fax, or streaming.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 SetPlayoutMode(
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const AudioPlayoutMode mode);
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//
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// PlayoutMode()
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// Get the current playout mode.
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//
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// Return value : The current playout mode.
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//
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AudioPlayoutMode PlayoutMode() const;
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//
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// NetworkStatistics()
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// Get the current network statistics from NetEQ.
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//
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// Output:
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// - statistics : The current network statistics.
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//
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// Return value : 0 if ok.
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// <0 if NetEQ returned an error.
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//
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WebRtc_Word32 NetworkStatistics(
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ACMNetworkStatistics* statistics) const;
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//
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// VADMode()
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// Get the current VAD Mode.
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//
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// Return value : The current VAD mode.
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//
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ACMVADMode VADMode() const;
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//
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// SetVADMode()
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// Set the VAD mode.
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//
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// Input:
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// - mode : The new VAD mode.
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//
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// Return value : 0 if ok.
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// -1 if an error occurred.
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//
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WebRtc_Word16 SetVADMode(
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const ACMVADMode mode);
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//
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// DecodeLock()
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// Get the decode lock used to protect decoder instances while decoding.
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//
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// Return value : Pointer to the decode lock.
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//
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RWLockWrapper* DecodeLock() const
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{
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return _decodeLock;
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}
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//
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// FlushBuffers()
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// Flushes the NetEQ packet and speech buffers.
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//
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// Return value : 0 if ok.
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// -1 if NetEQ returned an error.
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//
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WebRtc_Word32 FlushBuffers();
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//
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// RemoveCodec()
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// Removes a codec from the NetEQ codec database.
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//
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// Input:
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// - codecIdx : Codec to be removed.
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//
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// Return value : 0 if ok.
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// -1 if an error occurred.
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//
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WebRtc_Word16 RemoveCodec(
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WebRtcNetEQDecoder codecIdx,
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bool isStereo = false);
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//
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// SetBackgroundNoiseMode()
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// Set the mode of the background noise.
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//
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// Input:
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// - mode : an enumerator specifying the mode of the
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// background noise.
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//
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// Return value : 0 if succeeded,
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// -1 if failed to set the mode.
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//
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WebRtc_Word16 SetBackgroundNoiseMode(
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const ACMBackgroundNoiseMode mode);
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//
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// BackgroundNoiseMode()
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// return the mode of the background noise.
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//
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// Return value : The mode of background noise.
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//
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WebRtc_Word16 BackgroundNoiseMode(
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ACMBackgroundNoiseMode& mode);
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void SetUniqueId(
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WebRtc_Word32 id);
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WebRtc_Word32 PlayoutTimestamp(
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WebRtc_UWord32& timestamp);
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void SetReceivedStereo(
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bool receivedStereo);
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WebRtc_UWord8 NumSlaves();
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enum JB {masterJB = 0, slaveJB = 1};
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WebRtc_Word16 AddSlave(
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const WebRtcNetEQDecoder* usedCodecs,
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WebRtc_Word16 noOfCodecs);
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private:
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//
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// RTPPack()
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// Creates a Word16 RTP packet out of the payload data in Word16 and
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// a WebRtcRTPHeader.
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//
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// Input:
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// - payload : Payload to be packetized.
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// - payloadLengthW8 : Length of the payload in bytes.
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// - rtpInfo : RTP header struct.
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//
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// Output:
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// - rtpPacket : The RTP packet.
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//
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static void RTPPack(
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WebRtc_Word16* rtpPacket,
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const WebRtc_Word8* payload,
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const WebRtc_Word32 payloadLengthW8,
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const WebRtcRTPHeader& rtpInfo);
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void LogError(
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const char* neteqFuncName,
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const WebRtc_Word16 idx) const;
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WebRtc_Word16 InitByIdxSafe(
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const WebRtc_Word16 idx);
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// EnableVAD()
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// Enable VAD.
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//
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// Return value : 0 if ok.
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// -1 if an error occurred.
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//
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WebRtc_Word16 EnableVAD();
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WebRtc_Word16 EnableVADByIdxSafe(
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const WebRtc_Word16 idx);
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WebRtc_Word16 AllocatePacketBufferByIdxSafe(
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const WebRtcNetEQDecoder* usedCodecs,
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WebRtc_Word16 noOfCodecs,
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const WebRtc_Word16 idx);
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void* _inst[MAX_NUM_SLAVE_NETEQ + 1];
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void* _instMem[MAX_NUM_SLAVE_NETEQ + 1];
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WebRtc_Word16* _netEqPacketBuffer[MAX_NUM_SLAVE_NETEQ + 1];
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WebRtc_Word32 _id;
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float _currentSampFreqKHz;
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bool _avtPlayout;
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AudioPlayoutMode _playoutMode;
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CriticalSectionWrapper* _netEqCritSect;
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WebRtcVadInst* _ptrVADInst[MAX_NUM_SLAVE_NETEQ + 1];
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bool _vadStatus;
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ACMVADMode _vadMode;
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RWLockWrapper* _decodeLock;
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bool _isInitialized[MAX_NUM_SLAVE_NETEQ + 1];
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WebRtc_UWord8 _numSlaves;
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bool _receivedStereo;
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void* _masterSlaveInfo;
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AudioFrame::VADActivity _previousAudioActivity;
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WebRtc_Word32 _extraDelay;
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CriticalSectionWrapper* _callbackCritSect;
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};
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} //namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
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